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Questions tagged [sip-server]

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.

The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.

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Twilio SIP directly to phone

I have an IP phone but not a SIP Registry. Is it possible for me to set up my IP phone directly with Twilio SIP without having to depend on OpenPBX or FreePBX, just Twilio directly to my IP phone?
ecorvo
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Restcomm Web Trigger returns HTTP 401

I'm trying to use Web Trigger in a RVD project, but when I do the HTTP call this error appears: Error 401 running REST GET request: Unauthorized After some minutes under debug, I've noticed RvdController.java is trying to access the hardcoded path…
ghjansen
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Restcomm default username and password

Hi Guys I was trying run mobicents Telestax RestComm, and after I and the "mobicents-media-server\bin\run.bat" and the "Mobicents-RestComm-Jboss-AS7-7.2.2.616\bin\standalone.bat -c standalone-sip.xml" and both commands ran fine. and the I tried to…
Njabulo Nsibande
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OpenSIPS and Twilio Elastic Sip Trunking

I'm very very new to the world of VoIP, so please forgive my terminological inaccuracies. I have just installed ( on to a remote server ) the OpenSIPS software. I have provisioned a Twilio Elastic SIP Trunk which has it's Origination SIP URI set to…
Deano
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Unable to find the Internal port of the NAT mapping in the PJSIP/PJNATH library

I using the PJSIP/PJNATH port-punching library (for STUN/TURN/ICE) and want to create my own UDP-based transport over the punched-ports. I’m unable to find the data-structure that holds the internal port of the NAT mapping. The reason I need the…
auro
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Creating simple SIP service

Recently I've discovered Linphone which offers iOS SDK for creating VoIP calls. I would like to integrate it into my existing chat application and problem is that I need to make SIP service on my server which gives me a headaches. Is there any out…
Josip B.
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making SIP calls over TCP using PJSIP

I'm using a PJSIP's pjsua dialer (based on pjsua_app.c, PJSIP 2.0.1) with TCP transport and a SIP trunk to make calls to a mobile phone. The dialer registers with a SIP Server over TCP and also sends out INVITES over TCP. UDP transport is not being…
John Qualis
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PJSIP error pjsua_call_media_init

I had compiled the code from this link, also I am able to successfully register a account on sip server. But when I make a sip call then it gives a trap error in pjsua_call_media_init. I need help to understand the reason for error mentioned…
user2380738
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How to provision a test user in kamailio?

I have just (for the first time) compiled and installed kamailio, following this guide. For configuration, I am following the documentation here I am trying to test a new SIP user. I have created it with: » kamctl add test testpasswd The user is…
blueFast
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It is possible to use pjsip to connect to a 3CX Phone System over the 3CX tunnel?

I will split the question in some sub questions: Is the 3CX tunnel a true TCP SIP proxy? It is possible to use the pjsip to connect to it? How should I set up the account and connection data if this is possible? Let's consider the 3CX server is…
alinoz
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Custom Data (Text, Image, Voice, files...) over SIP

I am new to this board and I am also somewhat new to Androids. I am currently working with a research project for developing communities and we need to figure out a way to send custom data over SIP. Right now I am using an application called…
WiLD Net NapoNET
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maximum number of challenges to a UE in SIP?

How many times can a Registrar / Proxy challenge a UE with 401/407? If this UE is a malicious UE which keeps on sending random challenge responses, how can the Proxy / Registrar cease from honouring any more request, so that it can protect itself…
Radhakrishnan
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asteriskwin32 and WS

Can asteriskwin32 be installed on a windows service 2008? I am asking this because in my case i couldn't install it. I have a bash.exe error. I don;t know how to solve the problem If asteriskwin32 is not compatible with WS2008 are there other…
justmee
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How to generate big number of SIP requests

I need to test an application that processes SIP requests. For now, I want to test the performance of the application, so I need a way to generate a big number of SIP requests. I know there are tools for this (like SipP), but I don't know what is…
manash
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Unable to recieve SIP calls when app is in background

I am working on an iOS app that uses push notifications to wake up the app so I can receive SIP calls when app is not in foreground. When all is working, app gets a VOIP push notification while in background which wakes up the app and within 5…
Surbhi Arora
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